similar to: Update Mysql with DTMF

Hi Here is part of my extensions.conf exten => 8661231234,1,agi,dtmf.php When I dial this number, this is what I see in my asterisk console: -- Accepting AUTHENTICATED call from 198.22.67.70: > requested format = gsm, > requested prefs = (), > actual format = gsm, > host prefs = (gsm|ilbc|speex), > priority = mine -- Executing

while using iax and a soft phone, the sound of breathing comes through so clearly that it has started bothering me. Earlier I was amazed at the quality, but now feel it is irritating. Wondering if there is a way to cut it down. I am in the process of exploring using iax for a call center, but this sound of breathing is a disappointment. Thanks Hari

On 5/8/2024 7:53 AM, Alex Deucher wrote: > On Tue, May 7, 2024 at 2:32?PM Easwar Hariharan > <eahariha at linux.microsoft.com> wrote: >> >> On 5/3/2024 11:13 AM, Easwar Hariharan wrote: >>> I2C v7, SMBus 3.2, and I3C 1.1.1 specifications have replaced "master/slave" >>> with more appropriate terms. Inspired by and following on to Wolfram's

Hi, How do you transfer (using say blind transfer) a call to an extension that is currently busy on another call? You don't want the call to be transferred to voicemail, it must stay in 'hold' until the extension becomes available, and then immediately ring that phone. Thanks, Thomas

Is there a guide anywhere which runs through how to set up asterisk with mysql? I've looked and almost all the document misses out relevant information. Thanks Dan Journo -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050920/1ce45adb/attachment.htm

On 5/3/2024 11:13 AM, Easwar Hariharan wrote: > I2C v7, SMBus 3.2, and I3C 1.1.1 specifications have replaced "master/slave" > with more appropriate terms. Inspired by and following on to Wolfram's > series to fix drivers/i2c/[1], fix the terminology for users of > I2C_ALGOBIT bitbanging interface, now that the approved verbiage exists > in the specification. >

Hi, Simple description of the problem below. I have code coming into pruneEH as follows fn a { entry: call fn b ... for_cond: %i = phi [1, entry] [%x, for_body] cmp $i with someval cond-br for_body or for_exit for_body: ... $x = $i + 1 branch for_cond for_exit ... } PruneEH determines that the call to fn-b won't return. The code is modified thus. fn a { entry: call fn b unreachable insn

On Tue, May 7, 2024 at 2:32?PM Easwar Hariharan <eahariha at linux.microsoft.com> wrote: > > On 5/3/2024 11:13 AM, Easwar Hariharan wrote: > > I2C v7, SMBus 3.2, and I3C 1.1.1 specifications have replaced "master/slave" > > with more appropriate terms. Inspired by and following on to Wolfram's > > series to fix drivers/i2c/[1], fix the terminology for

Is it possible to get an internal dial tone when I call to my asterisk and enter password? I would like to call my line enter extension - password - and get internal dial tone. once I'm in I would like to dial based on what context permits, mostly long distance calls VOIP. I can not preset the extension to certain number as I don't know what number I will be dialing. -- #Joseph

I'm looking into solutions for providing a live stream of an event in Belgium [1] - for example, as follows: * Event --> mobile phone --> software answering machine --> Internet server * Event --> mobile phone --> VOIP --> Internet server The live stream should be available in a format so that people can listen to it using XMMS or similar software. Comments? Would

i have got a music file with extension mp3 and it is not workign with background() is there any way to convert the mp3 to gsm or any other codec? Kindest Muhammad Muzzamil Luqman -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050222/2ba5a4f0/attachment.htm

I have a client that has an old Merlin system. They would like to move to an Asterisk based system, however, with their existing system each phone is capable of displaying who is on the phone within there office. This is done by lighting a red light for each line(extension) that is in use. Has anyone been able to neatly create this feature? Perhaps an XML application can be written for the Cisco

I have added the following to a macro that is used for all extensions so a user can access voicemailmain by pressing * during the voicemail prompt ; check voicemail exten => a,1,voicemailmain(${macro_exten}) exten => a,2,hangup The behavior is a little weird, the * key is not recognized during the portion of the greeting where the extension number is being played back, after it is

I have Asterisk setup in a SOHO environment. I have 2 X100P cards at Zap/1 and Zap/2. I have 1 TDM400P card with Zap/3 - Zap/5. I have subscribed to callwaiting, callerid and calleridcallwaiting from Qwest on the 2 PSTN lines - Zap/1 and Zap/2. My problem is when I'm in an active call to the outside thru Zap/1 or Zap/2, I can't pickup the incoming callwaiting call. I can see the

Finally got everything up and run with the help of Manny Wise last night. So I am setting up my digital assistant and getting down to the task I need this box to perform the most. I need to have a custom app that I can call that will take me pressing 2 at the menu and have it transfer the call to a offsite phone number utilizing my Zap Trunk. I'm sure someone has done this already. Anyone want

I read the feature list of asterisk and I cannot see if it is possible to conference a call between extensions. Is it a supported feature of asterisk or is it done in the UA (ATA186 in my case) Here is what I try to do. phone-a -dial-> phone-b tap the cradle (flash on phone-a) phone-a -dial-> phone-c tap the cradle (flash on phone-a) Now I like all 3 phones in a conference call.

The subject says it all. A couple of my sons have very annoying friends that tend to call ALOT. I usually don't like to answer the phone but these kids keep calling back with in 2 minutes of calling. I'm sure someone else has this problem and maybe using * to do a callerID match and block? Even add logic that if they called so many times in an hour? Or in my case, make it a

The wiki says this about the MusicOnHold command: "Plays hold music specified by class. If omitted, the default music source for the channel will be used." http://www.voip-info.org/tiki-index.php?page=Asterisk%20cmd%20MusicOnHold How do I set the default music on hold class for the SIP channel ? I tried adding musiconhold=test to my sip.conf. musiconhold.conf looks like this:

hi, when i install the openssh rpm ["rpm -i --excludedocs openssh-3.4p1-105.i586.rpm"] i get an error saying cat: .//usr/share/doc/packages/openssh/README.SuSE: No such file or directory when i install the same with the --noscripts option, the installation goes through fine. That means one of the postinstall scripts of the openssh package expects a document to be installed by the

we have implemented teh following code for determinging the clustering model of a dataset. bicvals <- EMclust( hdata, 7) sumry1 <- summary(bicvals, hdata,7) # summary object for emclust() print(sumry1) This set of code gives the following output classification table: 1 2 3 4 5 6 7 1 1 1 4 1 1 1 which I think means there is 1 gene in the 1st cluster...1 gene in the 2nd cluster ,